The present invention relates to interpolating filters, and more particularly, to an interpolating filter and filtering technique that uses signal sample replication in lieu of zero insertion to achieve interpolation of digital signals.
The traditional technique of interpolating digital signals is to increase the sample rate of the digital signals by inserting zero-valued samples between the samples of the signal, then filtering the resulting signal with an interpolating filter. Prior art relating to the present invention is disclosed in the following patents. U.S. Pat. No. 4,109,110 issued to Gingell entitled "Digital-to-Analog Converter" teaches multiplying the sampling rate of a digital signal by repeating the sampled signal the required number of times, with reference to FIG. 6 and the description at col. 4, lines 12-22. The Gingell patent teaches the use of an interpolator as part of a process of increasing the data rate by a large factor for reproducing music from a compact disk, for example. The interpolator can be a simple sample and hold in the digital world according to the patent. There is no suggestion that a smoothing filter to remove the sample and hold artifacts be employed. Gingell does suggest that perhaps a simply constructed linear interpolator be used as being a little better than a sample and hold. The technique of Gingell follows the interpolator by other processing steps that increase the sample rate still farther. The digital to analog converter and the following analog filter are very simple as a result. The use of a filter as part of the interpolator that corrects the sample and hold frequency distortion is not suggested. Since there is no such filter, there is no consideration of the number of bits used in the quantization of the filter coefficients.
U.S. Pat. No. 4,270,026 issued to Shenoi et al. entitled "Interpolator apparatus for Increasing the Word Rate of a Digital Signal of the Type Employed in Digital Telephone Systems". This patent suggests using the traditional inserting of zero value samples to perform the increase in data rate for the digital signal. A recursive digital filter is used as the interpolator. The two pole recursive filter that is suggested will have very much less performance than the finite impulse response filter of the present invention. The present digital filter corrects for the digital sample and hold that I use while achieving the very sharp cutoff. Since this technique does not use a digital sample and hold, but a simple zero insert, there is no need to correct for the distortion, but the following filtering must use more accurate arithmetic. The subject of the Shenoi patent is a technique for implementing the recursive filter easily.
U.S. Pat. No. 5,075,880 issued to Moses et al. entitled "Method and Apparatus for Time Domain Interpolation of Digital Audio Signals" uses a traditional zero insertion technique to increase the sample rate. The filtering is performed by using Lagrangian or spline interpolation from a mathematical technique for interpolation. The patent argues that there is no need to worry about the frequency response of the result, a simple mathematical interpolation operation is good enough. Since there is no filter, there is no consideration of the accuracy of the filter coefficients required to implement the filters.
U.S. Pat. No. 4,209,771 issued to Miyata et at. entitled "Code Converting Method and System" implements a scheme for conversion to a high rate differential pulse code system using .+-.1 values. The conversion uses the usual DPCM approach of a linearly rising value at a high rate that can reach the value of the digital samples at the sample times. The result is an approximation to a carefully interpolated signal that may be acceptable in some applications. The distortions that occur are well known. As long as the signal changes slowly, the DPCM can keep up with it. When the signal changes rapidly with a large swing, the DPCM approximation will depart from the input signal, causing distortions in the signal and generating spurious frequencies that can cause interchannel interference when there is more than one signal in the bandwidth of the digital samples.
U.S. Pat. No. 5,126,737 issued to Torii entitled "Method for Converting a Digital Signal into Another Digital Signal Having a Different Sampling Frequency" uses interpolating to produce an output sample rate that is nearly the same as the input sample rate. The technique proposed by Torii is to increase the sample rate by a large factor, then downsample to the required rate. The fact that the output samples can drift between the samples even at the very high rate is accounted for by a linear interpolator at the very high rate. The patent suggests using a simple linear interpolation between the two nearest samples. A phase-locked loop is used to determine how far between the two samples the output sample is supposed to occur. The step up to the high sample rate includes the possibility of using a sample and hold. Any technique for producing the high sample rate is acceptable for the purposes of this patent. There is no consideration of the accuracy of the arithmetic involved. There is no consideration of the possibility of correcting for the sample and hold in the filtering process.
U.S. Pat. No. 4,630,034 issued to Takahashi entitled "Sampling Frequency Converting Apparatus" provides a description of a way to change television signals from one format to another. The technique uses a buffering scheme to collect "M" samples of the input, then uses standard digital filtering techniques to produce "N" output samples. The ratio of M/N provides the conversion. The filtering uses large number of sets of coefficients to generate filters that are equivalent to upsampling using zero insertion by a large factor, filtering, then downsampling.
U.S. Pat. No. 4,460,890 issued to Busby entitled "Direct Digital to Digital Sampling Rate Conversion, Method and Apparatus" teaches something similar to Torii and to Moses above. The filters used are simple finite impulse response falters applied to signals whose sample rate is increased by effectively inserting zeroes. The Busby patent suggests interpolating the high sample rate signal with a polynomial interpolator instead of a spline or Lagrangian interpolation. Busby does suggest calculating only those higher rate samples needed in the interpolation of the output points. There is no consideration of the accuracy of the arithmetic used.
U.S. Pat. No. 4,903,019 issued to Ito entitled "Sampling Frequency converter for Converting a Lower Sampling Frequency to a Higher Sampling Frequency and a Method Therefor: suggests using a nearly traditional interpolation technique. Instead of inserting zeroes between samples, this patent suggests inserting zeroes in a block at the end of a group of input samples. The number of zeroes inserted will be those required to pad a block of samples from M samples in the input block to N samples for the output block. The falter uses different sets of coefficients for each of the output points. The patent does not suggest any particular filters, just those that are well known in the industry. The patent does not consider the arithmetic accuracy. The scheme is equivalent to the scheme of inserting zeroes to upsample, then filtering to downsample to the new rate.
Thus, from the above, several of the schemes discloses in the prior art patents use the traditional insert zeroes technique for upsampling before filtering. Others suggest a digital sample and hold that is similar to one used in the present invention. Further, none of the above patents are concerned with a finite impulse response filter with a limited number of bits for coefficients. The accuracy of the arithmetic in the recursive interpolating filter is addressed in the present invention and is not in the prior art patents. The prior art techniques that build a DPCM signal reduce the output to a very low number of bits at a high frequency, but do not consider the accuracy of the arithmetic in interpolating filters that are used in intermediate steps.
Therefore, it is an objective of the present invention to provide for an improved filtering technique that uses signal sample replication with a correction filter in lieu of zero insertion to achieve interpolation of digital signals.